The number of voice communication systems has grown significantly as technology has advanced. Accordingly, bandwidth over which voice data is transmitted and received is at a premium and technological advances for conserving bandwidth are desired. For example, Global System for Mobile Communications (“GSM”) standards as well as non-GSM standards provide for transmission of data at different maximum bit rates. Accordingly, transmitting data at below the maximum bit rate for a particular standard is imperative. Adaptive multi-rate (“AMR”) codecs operate according to encoding and decoding schemes that can reduce the required bandwidth for representing data. AMR codecs can be tailored to provide encoding and decoding at a number of different bit rates based on factors such as the channel conditions, system requirements or the like. The conventional AMR protocol is disclosed in “3GPP TS 26.101 v7.0.0 (2007-06) 3rd Generation Partnership Project; Technical Specification Group Services and System Aspects; Mandatory speech codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec frame structure (Release 7),” which is incorporated herein by reference in its entirety. Unfortunately, conventional AMR protocols utilize unnecessary bandwidth thereby reducing the potential bandwidth efficiency of voice communications systems.
Further, improvements in technology have also provided opportunities for compromising the security and/or integrity of voice communications. Unfortunately, security protocols to protect voice data typically disadvantageously result in an increase in the overhead for the voice communications and a loss of bandwidth efficiency. As such, there is a desire for systems and methods for improving the bandwidth efficiency of AMR codecs that process voice data thereby allowing for a greater amount of bandwidth for the provisioning of security.